mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-12-04 23:46:43 +00:00
98 lines
3.4 KiB
C
98 lines
3.4 KiB
C
/*
|
|
* WebRTC Audio Processing Elements
|
|
*
|
|
* Copyright 2016 Collabora Ltd
|
|
* @author: Nicolas Dufresne <nicolas.dufresne@collabora.com>
|
|
*
|
|
* This library is free software; you can redistribute it and/or
|
|
* modify it under the terms of the GNU Lesser General Public
|
|
* License as published by the Free Software Foundation; either
|
|
* version 2.1 of the License, or (at your option) any later version.
|
|
*
|
|
* This library is distributed in the hope that it will be useful,
|
|
* but WITHOUT ANY WARRANTY; without even the implied warranty of
|
|
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
|
|
* Lesser General Public License for more details.
|
|
*
|
|
* You should have received a copy of the GNU Lesser General Public
|
|
* License along with this library; if not, write to the Free Software
|
|
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
|
|
*
|
|
*/
|
|
|
|
#ifndef __GST_WEBRTC_ECHO_PROBE_H__
|
|
#define __GST_WEBRTC_ECHO_PROBE_H__
|
|
|
|
#include <gst/gst.h>
|
|
#include <gst/base/gstadapter.h>
|
|
#include <gst/base/gstbasetransform.h>
|
|
#include <gst/audio/audio.h>
|
|
|
|
#ifndef GST_USE_UNSTABLE_API
|
|
#define GST_USE_UNSTABLE_API
|
|
#endif
|
|
#include <gst/audio/gstplanaraudioadapter.h>
|
|
|
|
G_BEGIN_DECLS
|
|
|
|
#define GST_TYPE_WEBRTC_ECHO_PROBE (gst_webrtc_echo_probe_get_type())
|
|
#define GST_WEBRTC_ECHO_PROBE(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_WEBRTC_ECHO_PROBE,GstWebrtcEchoProbe))
|
|
#define GST_IS_WEBRTC_ECHO_PROBE(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_WEBRTC_ECHO_PROBE))
|
|
#define GST_WEBRTC_ECHO_PROBE_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass) ,GST_TYPE_WEBRTC_ECHO_PROBE,GstWebrtcEchoProbeClass))
|
|
#define GST_IS_WEBRTC_ECHO_PROBE_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,GST_TYPE_WEBRTC_ECHO_PROBE))
|
|
#define GST_WEBRTC_ECHO_PROBE_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_WEBRTC_ECHO_PROBE,GstWebrtcEchoProbeClass))
|
|
|
|
#define GST_WEBRTC_ECHO_PROBE_LOCK(obj) g_mutex_lock (&GST_WEBRTC_ECHO_PROBE (obj)->lock)
|
|
#define GST_WEBRTC_ECHO_PROBE_UNLOCK(obj) g_mutex_unlock (&GST_WEBRTC_ECHO_PROBE (obj)->lock)
|
|
|
|
typedef struct _GstWebrtcEchoProbe GstWebrtcEchoProbe;
|
|
typedef struct _GstWebrtcEchoProbeClass GstWebrtcEchoProbeClass;
|
|
|
|
/**
|
|
* GstWebrtcEchoProbe:
|
|
*
|
|
* The adder object structure.
|
|
*/
|
|
struct _GstWebrtcEchoProbe
|
|
{
|
|
GstAudioFilter parent;
|
|
|
|
/* This lock is required as the DSP may need to lock itself using it's
|
|
* object lock and also lock the probe. The natural order for the DSP is
|
|
* to lock the DSP and then the echo probe. If we where using the probe
|
|
* object lock, we'd be racing with GstBin which will lock sink to src,
|
|
* and may accidentally reverse the order. */
|
|
GMutex lock;
|
|
|
|
/* Protected by the lock */
|
|
GstAudioInfo info;
|
|
guint period_size;
|
|
guint period_samples;
|
|
GstClockTime latency;
|
|
gint delay;
|
|
gboolean interleaved;
|
|
|
|
GstSegment segment;
|
|
GstAdapter *adapter;
|
|
GstPlanarAudioAdapter *padapter;
|
|
|
|
/* Private */
|
|
gboolean acquired;
|
|
};
|
|
|
|
struct _GstWebrtcEchoProbeClass
|
|
{
|
|
GstAudioFilterClass parent_class;
|
|
};
|
|
|
|
GType gst_webrtc_echo_probe_get_type (void);
|
|
|
|
GST_ELEMENT_REGISTER_DECLARE (webrtcechoprobe);
|
|
|
|
GstWebrtcEchoProbe *gst_webrtc_acquire_echo_probe (const gchar * name);
|
|
void gst_webrtc_release_echo_probe (GstWebrtcEchoProbe * probe);
|
|
gint gst_webrtc_echo_probe_read (GstWebrtcEchoProbe * self,
|
|
GstClockTime rec_time, gpointer frame, GstBuffer ** buf);
|
|
|
|
G_END_DECLS
|
|
#endif /* __GST_WEBRTC_ECHO_PROBE_H__ */
|